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webrtc: improve matching on the correct jitterbuffer

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The mapping between an RTP session and the SDP m= line is not always the
same, especially when BUNDLEing is used.

This causes a failure in a specific case where if when bundling,
if mline 0 is a data channel, and mline 1 an audio/video section,
then retrieving the transceiver at mline 0 (rtp session used) will fail
and cause an assertion.

This fix is actually potentially a regression for cases where the remote
part does not provide the a=ssrc: media level SDP attributes as is now
becoming common, especially when simulcast is involved.

The correct fix actually requires reading out header extensions as used
with bundle for signalling in the actual data, what media and therefore
transceiver is being used.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2467>
This commit is contained in:
Matthew Waters 2021-05-06 22:22:12 +10:00 committed by GStreamer Marge Bot
parent fbf6bfd4d8
commit 18314764fc
1 changed files with 32 additions and 13 deletions
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43
ext/webrtc/gstwebrtcbin.c
View file

@ -6412,28 +6412,47 @@ static void
on_rtpbin_new_jitterbuffer (GstElement * rtpbin, GstElement * jitterbuffer,
guint session_id, guint ssrc, GstWebRTCBin * webrtc)
{
WebRTCTransceiver *trans;
TransportStream *stream;
guint i;
trans = (WebRTCTransceiver *) _find_transceiver (webrtc, &session_id,
(FindTransceiverFunc) transceiver_match_for_mline);
PC_LOCK (webrtc);
GST_INFO_OBJECT (webrtc, "new jitterbuffer %" GST_PTR_FORMAT " for "
"session %u ssrc %u", jitterbuffer, session_id, ssrc);
if (trans) {
/* We don't set do-retransmission on rtpbin as we want per-session control */
g_object_set (jitterbuffer, "do-retransmission",
WEBRTC_TRANSCEIVER (trans)->do_nack, NULL);
if (!(stream = _find_transport_for_session (webrtc, session_id))) {
g_warn_if_reached ();
goto out;
}
for (i = 0; i < trans->stream->remote_ssrcmap->len; i++) {
SsrcMapItem *item = g_ptr_array_index (trans->stream->remote_ssrcmap, i);
/* XXX: this will fail with no ssrc in the remote sdp as used with e.g. simulcast
* newer SDP versions from chrome/firefox */
for (i = 0; i < stream->remote_ssrcmap->len; i++) {
SsrcMapItem *item = g_ptr_array_index (stream->remote_ssrcmap, i);
if (item->ssrc == ssrc) {
GstWebRTCRTPTransceiver *trans;
gboolean do_nack;
trans = _find_transceiver_for_mline (webrtc, item->media_idx);
if (!trans) {
g_warn_if_reached ();
break;
}
do_nack = WEBRTC_TRANSCEIVER (trans)->do_nack;
/* We don't set do-retransmission on rtpbin as we want per-session control */
GST_LOG_OBJECT (webrtc, "setting do-nack=%s for transceiver %"
GST_PTR_FORMAT " with transport %" GST_PTR_FORMAT
" rtp session %u ssrc %u", do_nack ? "true" : "false", trans, stream,
session_id, ssrc);
g_object_set (jitterbuffer, "do-retransmission", do_nack, NULL);
g_weak_ref_set (&item->rtpjitterbuffer, jitterbuffer);
break;
}
}
} else {
g_assert_not_reached ();
}
out:
PC_UNLOCK (webrtc);
}
static void